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FFmpeg/libavcodec/alac.c
Michael Niedermayer faba79e080 Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  mxfdec: Include FF_INPUT_BUFFER_PADDING_SIZE when allocating extradata.
  H.264: tweak some other x86 asm for Atom
  probe: Fix insane flow control.
  mpegts: remove invalid error check
  s302m: use nondeprecated audio sample format API
  lavc: use designated initialisers for all codecs.
  x86: cabac: add operand size suffixes missing from 6c32576

Conflicts:
	libavcodec/ac3enc_float.c
	libavcodec/flacenc.c
	libavcodec/frwu.c
	libavcodec/pictordec.c
	libavcodec/qtrleenc.c
	libavcodec/v210enc.c
	libavcodec/wmv2dec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-07-30 06:46:08 +02:00

691 lines
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/*
* ALAC (Apple Lossless Audio Codec) decoder
* Copyright (c) 2005 David Hammerton
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ALAC (Apple Lossless Audio Codec) decoder
* @author 2005 David Hammerton
* @see http://crazney.net/programs/itunes/alac.html
*
* Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
* passed through the extradata[_size] fields. This atom is tacked onto
* the end of an 'alac' stsd atom and has the following format:
* bytes 0-3 atom size (0x24), big-endian
* bytes 4-7 atom type ('alac', not the 'alac' tag from start of stsd)
* bytes 8-35 data bytes needed by decoder
*
* Extradata:
* 32bit size
* 32bit tag (=alac)
* 32bit zero?
* 32bit max sample per frame
* 8bit ?? (zero?)
* 8bit sample size
* 8bit history mult
* 8bit initial history
* 8bit kmodifier
* 8bit channels?
* 16bit ??
* 32bit max coded frame size
* 32bit bitrate?
* 32bit samplerate
*/
#include "avcodec.h"
#include "get_bits.h"
#include "bytestream.h"
#include "unary.h"
#include "mathops.h"
#define ALAC_EXTRADATA_SIZE 36
#define MAX_CHANNELS 2
typedef struct {
AVCodecContext *avctx;
GetBitContext gb;
int numchannels;
int bytespersample;
/* buffers */
int32_t *predicterror_buffer[MAX_CHANNELS];
int32_t *outputsamples_buffer[MAX_CHANNELS];
int32_t *wasted_bits_buffer[MAX_CHANNELS];
/* stuff from setinfo */
uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */
uint8_t setinfo_sample_size; /* 0x10 */
uint8_t setinfo_rice_historymult; /* 0x28 */
uint8_t setinfo_rice_initialhistory; /* 0x0a */
uint8_t setinfo_rice_kmodifier; /* 0x0e */
/* end setinfo stuff */
int wasted_bits;
} ALACContext;
static void allocate_buffers(ALACContext *alac)
{
int chan;
for (chan = 0; chan < MAX_CHANNELS; chan++) {
alac->predicterror_buffer[chan] =
av_malloc(alac->setinfo_max_samples_per_frame * 4);
alac->outputsamples_buffer[chan] =
av_malloc(alac->setinfo_max_samples_per_frame * 4);
alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4);
}
}
static int alac_set_info(ALACContext *alac)
{
const unsigned char *ptr = alac->avctx->extradata;
ptr += 4; /* size */
ptr += 4; /* alac */
ptr += 4; /* 0 ? */
if(AV_RB32(ptr) >= UINT_MAX/4){
av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
return -1;
}
/* buffer size / 2 ? */
alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
ptr++; /* ??? */
alac->setinfo_sample_size = *ptr++;
if (alac->setinfo_sample_size > 32) {
av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
return -1;
}
alac->setinfo_rice_historymult = *ptr++;
alac->setinfo_rice_initialhistory = *ptr++;
alac->setinfo_rice_kmodifier = *ptr++;
ptr++; /* channels? */
bytestream_get_be16(&ptr); /* ??? */
bytestream_get_be32(&ptr); /* max coded frame size */
bytestream_get_be32(&ptr); /* bitrate ? */
bytestream_get_be32(&ptr); /* samplerate */
allocate_buffers(alac);
return 0;
}
static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){
/* read x - number of 1s before 0 represent the rice */
int x = get_unary_0_9(gb);
if (x > 8) { /* RICE THRESHOLD */
/* use alternative encoding */
x = get_bits(gb, readsamplesize);
} else {
if (k >= limit)
k = limit;
if (k != 1) {
int extrabits = show_bits(gb, k);
/* multiply x by 2^k - 1, as part of their strange algorithm */
x = (x << k) - x;
if (extrabits > 1) {
x += extrabits - 1;
skip_bits(gb, k);
} else
skip_bits(gb, k - 1);
}
}
return x;
}
static void bastardized_rice_decompress(ALACContext *alac,
int32_t *output_buffer,
int output_size,
int readsamplesize, /* arg_10 */
int rice_initialhistory, /* arg424->b */
int rice_kmodifier, /* arg424->d */
int rice_historymult, /* arg424->c */
int rice_kmodifier_mask /* arg424->e */
)
{
int output_count;
unsigned int history = rice_initialhistory;
int sign_modifier = 0;
for (output_count = 0; output_count < output_size; output_count++) {
int32_t x;
int32_t x_modified;
int32_t final_val;
/* standard rice encoding */
int k; /* size of extra bits */
/* read k, that is bits as is */
k = av_log2((history >> 9) + 3);
x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize);
x_modified = sign_modifier + x;
final_val = (x_modified + 1) / 2;
if (x_modified & 1) final_val *= -1;
output_buffer[output_count] = final_val;
sign_modifier = 0;
/* now update the history */
history += x_modified * rice_historymult
- ((history * rice_historymult) >> 9);
if (x_modified > 0xffff)
history = 0xffff;
/* special case: there may be compressed blocks of 0 */
if ((history < 128) && (output_count+1 < output_size)) {
int k;
unsigned int block_size;
sign_modifier = 1;
k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);
block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);
if (block_size > 0) {
if(block_size >= output_size - output_count){
av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count);
block_size= output_size - output_count - 1;
}
memset(&output_buffer[output_count+1], 0, block_size * 4);
output_count += block_size;
}
if (block_size > 0xffff)
sign_modifier = 0;
history = 0;
}
}
}
static inline int sign_only(int v)
{
return v ? FFSIGN(v) : 0;
}
static void predictor_decompress_fir_adapt(int32_t *error_buffer,
int32_t *buffer_out,
int output_size,
int readsamplesize,
int16_t *predictor_coef_table,
int predictor_coef_num,
int predictor_quantitization)
{
int i;
/* first sample always copies */
*buffer_out = *error_buffer;
if (!predictor_coef_num) {
if (output_size <= 1)
return;
memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4);
return;
}
if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */
/* second-best case scenario for fir decompression,
* error describes a small difference from the previous sample only
*/
if (output_size <= 1)
return;
for (i = 0; i < output_size - 1; i++) {
int32_t prev_value;
int32_t error_value;
prev_value = buffer_out[i];
error_value = error_buffer[i+1];
buffer_out[i+1] =
sign_extend((prev_value + error_value), readsamplesize);
}
return;
}
/* read warm-up samples */
if (predictor_coef_num > 0)
for (i = 0; i < predictor_coef_num; i++) {
int32_t val;
val = buffer_out[i] + error_buffer[i+1];
val = sign_extend(val, readsamplesize);
buffer_out[i+1] = val;
}
/* 4 and 8 are very common cases (the only ones i've seen). these
* should be unrolled and optimized
*/
/* general case */
if (predictor_coef_num > 0) {
for (i = predictor_coef_num + 1; i < output_size; i++) {
int j;
int sum = 0;
int outval;
int error_val = error_buffer[i];
for (j = 0; j < predictor_coef_num; j++) {
sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) *
predictor_coef_table[j];
}
outval = (1 << (predictor_quantitization-1)) + sum;
outval = outval >> predictor_quantitization;
outval = outval + buffer_out[0] + error_val;
outval = sign_extend(outval, readsamplesize);
buffer_out[predictor_coef_num+1] = outval;
if (error_val > 0) {
int predictor_num = predictor_coef_num - 1;
while (predictor_num >= 0 && error_val > 0) {
int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
int sign = sign_only(val);
predictor_coef_table[predictor_num] -= sign;
val *= sign; /* absolute value */
error_val -= ((val >> predictor_quantitization) *
(predictor_coef_num - predictor_num));
predictor_num--;
}
} else if (error_val < 0) {
int predictor_num = predictor_coef_num - 1;
while (predictor_num >= 0 && error_val < 0) {
int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
int sign = - sign_only(val);
predictor_coef_table[predictor_num] -= sign;
val *= sign; /* neg value */
error_val -= ((val >> predictor_quantitization) *
(predictor_coef_num - predictor_num));
predictor_num--;
}
}
buffer_out++;
}
}
}
static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
int16_t *buffer_out,
int numchannels, int numsamples,
uint8_t interlacing_shift,
uint8_t interlacing_leftweight)
{
int i;
if (numsamples <= 0)
return;
/* weighted interlacing */
if (interlacing_leftweight) {
for (i = 0; i < numsamples; i++) {
int32_t a, b;
a = buffer[0][i];
b = buffer[1][i];
a -= (b * interlacing_leftweight) >> interlacing_shift;
b += a;
buffer_out[i*numchannels] = b;
buffer_out[i*numchannels + 1] = a;
}
return;
}
/* otherwise basic interlacing took place */
for (i = 0; i < numsamples; i++) {
int16_t left, right;
left = buffer[0][i];
right = buffer[1][i];
buffer_out[i*numchannels] = left;
buffer_out[i*numchannels + 1] = right;
}
}
static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS],
int32_t *buffer_out,
int32_t *wasted_bits_buffer[MAX_CHANNELS],
int wasted_bits,
int numchannels, int numsamples,
uint8_t interlacing_shift,
uint8_t interlacing_leftweight)
{
int i;
if (numsamples <= 0)
return;
/* weighted interlacing */
if (interlacing_leftweight) {
for (i = 0; i < numsamples; i++) {
int32_t a, b;
a = buffer[0][i];
b = buffer[1][i];
a -= (b * interlacing_leftweight) >> interlacing_shift;
b += a;
if (wasted_bits) {
b = (b << wasted_bits) | wasted_bits_buffer[0][i];
a = (a << wasted_bits) | wasted_bits_buffer[1][i];
}
buffer_out[i * numchannels] = b << 8;
buffer_out[i * numchannels + 1] = a << 8;
}
} else {
for (i = 0; i < numsamples; i++) {
int32_t left, right;
left = buffer[0][i];
right = buffer[1][i];
if (wasted_bits) {
left = (left << wasted_bits) | wasted_bits_buffer[0][i];
right = (right << wasted_bits) | wasted_bits_buffer[1][i];
}
buffer_out[i * numchannels] = left << 8;
buffer_out[i * numchannels + 1] = right << 8;
}
}
}
static int alac_decode_frame(AVCodecContext *avctx,
void *outbuffer, int *outputsize,
AVPacket *avpkt)
{
const uint8_t *inbuffer = avpkt->data;
int input_buffer_size = avpkt->size;
ALACContext *alac = avctx->priv_data;
int channels;
unsigned int outputsamples;
int hassize;
unsigned int readsamplesize;
int isnotcompressed;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
/* short-circuit null buffers */
if (!inbuffer || !input_buffer_size)
return -1;
init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
channels = get_bits(&alac->gb, 3) + 1;
if (channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
MAX_CHANNELS);
return -1;
}
/* 2^result = something to do with output waiting.
* perhaps matters if we read > 1 frame in a pass?
*/
skip_bits(&alac->gb, 4);
skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */
/* the output sample size is stored soon */
hassize = get_bits1(&alac->gb);
alac->wasted_bits = get_bits(&alac->gb, 2) << 3;
/* whether the frame is compressed */
isnotcompressed = get_bits1(&alac->gb);
if (hassize) {
/* now read the number of samples as a 32bit integer */
outputsamples = get_bits_long(&alac->gb, 32);
if(outputsamples > alac->setinfo_max_samples_per_frame){
av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame);
return -1;
}
} else
outputsamples = alac->setinfo_max_samples_per_frame;
switch (alac->setinfo_sample_size) {
case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
alac->bytespersample = channels << 1;
break;
case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
alac->bytespersample = channels << 2;
break;
default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n",
alac->setinfo_sample_size);
return -1;
}
if(outputsamples > *outputsize / alac->bytespersample){
av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
return -1;
}
*outputsize = outputsamples * alac->bytespersample;
readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1;
if (readsamplesize > MIN_CACHE_BITS) {
av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
return -1;
}
if (!isnotcompressed) {
/* so it is compressed */
int16_t predictor_coef_table[MAX_CHANNELS][32];
int predictor_coef_num[MAX_CHANNELS];
int prediction_type[MAX_CHANNELS];
int prediction_quantitization[MAX_CHANNELS];
int ricemodifier[MAX_CHANNELS];
int i, chan;
interlacing_shift = get_bits(&alac->gb, 8);
interlacing_leftweight = get_bits(&alac->gb, 8);
for (chan = 0; chan < channels; chan++) {
prediction_type[chan] = get_bits(&alac->gb, 4);
prediction_quantitization[chan] = get_bits(&alac->gb, 4);
ricemodifier[chan] = get_bits(&alac->gb, 3);
predictor_coef_num[chan] = get_bits(&alac->gb, 5);
/* read the predictor table */
for (i = 0; i < predictor_coef_num[chan]; i++)
predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
}
if (alac->wasted_bits) {
int i, ch;
for (i = 0; i < outputsamples; i++) {
for (ch = 0; ch < channels; ch++)
alac->wasted_bits_buffer[ch][i] = get_bits(&alac->gb, alac->wasted_bits);
}
}
for (chan = 0; chan < channels; chan++) {
bastardized_rice_decompress(alac,
alac->predicterror_buffer[chan],
outputsamples,
readsamplesize,
alac->setinfo_rice_initialhistory,
alac->setinfo_rice_kmodifier,
ricemodifier[chan] * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
if (prediction_type[chan] == 0) {
/* adaptive fir */
predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
alac->outputsamples_buffer[chan],
outputsamples,
readsamplesize,
predictor_coef_table[chan],
predictor_coef_num[chan],
prediction_quantitization[chan]);
} else {
av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
/* I think the only other prediction type (or perhaps this is
* just a boolean?) runs adaptive fir twice.. like:
* predictor_decompress_fir_adapt(predictor_error, tempout, ...)
* predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
* little strange..
*/
}
}
} else {
/* not compressed, easy case */
int i, chan;
if (alac->setinfo_sample_size <= 16) {
for (i = 0; i < outputsamples; i++)
for (chan = 0; chan < channels; chan++) {
int32_t audiobits;
audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);
alac->outputsamples_buffer[chan][i] = audiobits;
}
} else {
for (i = 0; i < outputsamples; i++) {
for (chan = 0; chan < channels; chan++) {
alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb,
alac->setinfo_sample_size);
alac->outputsamples_buffer[chan][i] = sign_extend(alac->outputsamples_buffer[chan][i],
alac->setinfo_sample_size);
}
}
}
alac->wasted_bits = 0;
interlacing_shift = 0;
interlacing_leftweight = 0;
}
if (get_bits(&alac->gb, 3) != 7)
av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
switch(alac->setinfo_sample_size) {
case 16:
if (channels == 2) {
reconstruct_stereo_16(alac->outputsamples_buffer,
(int16_t*)outbuffer,
alac->numchannels,
outputsamples,
interlacing_shift,
interlacing_leftweight);
} else {
int i;
for (i = 0; i < outputsamples; i++) {
((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
}
}
break;
case 24:
if (channels == 2) {
decorrelate_stereo_24(alac->outputsamples_buffer,
outbuffer,
alac->wasted_bits_buffer,
alac->wasted_bits,
alac->numchannels,
outputsamples,
interlacing_shift,
interlacing_leftweight);
} else {
int i;
for (i = 0; i < outputsamples; i++)
((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
}
break;
}
if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
return input_buffer_size;
}
static av_cold int alac_decode_init(AVCodecContext * avctx)
{
ALACContext *alac = avctx->priv_data;
alac->avctx = avctx;
alac->numchannels = alac->avctx->channels;
/* initialize from the extradata */
if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
ALAC_EXTRADATA_SIZE);
return -1;
}
if (alac_set_info(alac)) {
av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
return -1;
}
return 0;
}
static av_cold int alac_decode_close(AVCodecContext *avctx)
{
ALACContext *alac = avctx->priv_data;
int chan;
for (chan = 0; chan < MAX_CHANNELS; chan++) {
av_freep(&alac->predicterror_buffer[chan]);
av_freep(&alac->outputsamples_buffer[chan]);
av_freep(&alac->wasted_bits_buffer[chan]);
}
return 0;
}
AVCodec ff_alac_decoder = {
.name = "alac",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ALAC,
.priv_data_size = sizeof(ALACContext),
.init = alac_decode_init,
.close = alac_decode_close,
.decode = alac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
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